The present invention relates to internet telephony techniques, and more particular, to a method for determining voice quality in a voice call over a packet network and rerouting the voice call accordingly.
With the widespread of broadband Internet connections and the development of Internet telephony techniques, more and more calls are made over the Internet to take advantage of low prices. Though the users understand that they may have to sacrifice some voice quality, voice quality still needs to be kept at an acceptable level.
Currently, the objective methods for measuring voice quality of a voice call over a packet network such as the Internet using a protocol such as VOIP or SIP are mainly categorized into three types: 1) intrusive method: in this method, a reference signal is injected on one side of the network and collected on the other. An assessment of the quality of the network can be made by comparing any differences between the original and transmitted signals. This method gives a true end-to-end quality assessment, however, it is intrusive, requires a receiver to collect the transmitted signal, and does not account for network latency; 2) signal-based non-intrusive method: in this method, the voice quality is assessed by “listening” to ordinary phone calls and applying complex speech pattern recognition. This method is non-intrusive, and gives true end-to-end measurements. However, it does not account for network delay either. Moreover, it is impractical for a large-size fully-meshed network; and 3) parameter-based non-intrusive method: this method assesses voice quality by collecting impairment information from ordinary phone calls on the packet network. It uses the known E-model formula to derive the quality of a call. This method does not give true end-to-end results but only gives the voice quality of the packet network portion of the phone call.
The present invention is directed to the method of the third type as explained above, i.e., the parameter-based non-intrusive method.